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Spoon - Revenge !
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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My Morning Jacket - One Big Holiday
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Zap Mama - Wadidyusay ?
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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David Byrne - My Fair Lady
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Beastie Boys - Now Get Busy
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Granite de l’Aber Ildut
9 septembre 2011, par
Mis à jour : Septembre 2011
Langue : français
Type : Texte
Autres articles (31)
-
Librairies et binaires spécifiques au traitement vidéo et sonore
31 janvier 2010, parLes logiciels et librairies suivantes sont utilisées par SPIPmotion d’une manière ou d’une autre.
Binaires obligatoires FFMpeg : encodeur principal, permet de transcoder presque tous les types de fichiers vidéo et sonores dans les formats lisibles sur Internet. CF ce tutoriel pour son installation ; Oggz-tools : outils d’inspection de fichiers ogg ; Mediainfo : récupération d’informations depuis la plupart des formats vidéos et sonores ;
Binaires complémentaires et facultatifs flvtool2 : (...) -
Support audio et vidéo HTML5
10 avril 2011MediaSPIP utilise les balises HTML5 video et audio pour la lecture de documents multimedia en profitant des dernières innovations du W3C supportées par les navigateurs modernes.
Pour les navigateurs plus anciens, le lecteur flash Flowplayer est utilisé.
Le lecteur HTML5 utilisé a été spécifiquement créé pour MediaSPIP : il est complètement modifiable graphiquement pour correspondre à un thème choisi.
Ces technologies permettent de distribuer vidéo et son à la fois sur des ordinateurs conventionnels (...) -
De l’upload à la vidéo finale [version standalone]
31 janvier 2010, parLe chemin d’un document audio ou vidéo dans SPIPMotion est divisé en trois étapes distinctes.
Upload et récupération d’informations de la vidéo source
Dans un premier temps, il est nécessaire de créer un article SPIP et de lui joindre le document vidéo "source".
Au moment où ce document est joint à l’article, deux actions supplémentaires au comportement normal sont exécutées : La récupération des informations techniques des flux audio et video du fichier ; La génération d’une vignette : extraction d’une (...)
Sur d’autres sites (5104)
-
Discord Voice Bot cannot play the audio file
7 avril 2023, par Jakub NawrockiI tried to write a bot that will join the voice channel and play a audio at 20:00.


Currently the bot joins the channel, but immediately after that it disconnects without making a single sound with this message :


2023-04-07 17:58:01 INFO discord.player ffmpeg process 18258 has not terminated. Waiting to terminate... 2023-04-07 17:58:01 INFO discord.player ffmpeg process 18258 should have terminated with a return code of -9. 2023-04-07 17:58:01 INFO discord.voice_client The voice handshake is being terminated for Channel ID 1093533451778523241 (Guild ID 1093533451778523237) 2023-04-07 17:58:01 INFO discord.voice_client Disconnecting from voice normally, close code 1000. Audio file loaded: Audio could not be played:

Code :


import discord
import asyncio
import datetime

TOKEN = 'TOKEN HERE' 
CHANNEL_ID = CHANNEL ID HERE

client = discord.Client(intents=discord.Intents.all())

async def play_sound(voice_client):
 try:
 source = discord.FFmpegPCMAudio('audio.mp3')
 print(f"Audio file loaded: {source}")
 voice_client.play(source)
 while voice_client.is_playing():
 await asyncio.sleep(1)
 except Exception as e:
 print(f"Audio could not be played: {e}")

@client.event
async def on_ready():
 print('Bot is ready')
 now = datetime.datetime.now()
 target_time = datetime.time(hour=20, minute=00)
 if now.time() >= target_time:
 print(f"Current time: {now.time()}. Bot did not join channel.")
 return
 else:
 print(f"Current time: {now.time()}. Bot has joined at {target_time}.")
 await asyncio.sleep((datetime.datetime.combine(datetime.date.today(), target_time) - now).total_seconds())
 channel = client.get_channel(CHANNEL_ID)
 if channel is not None:
 try:
 voice_client = await channel.connect()
 print(f'{client.user} joined voice chat.')
 await asyncio.sleep(1)
 await play_sound(voice_client)
 await voice_client.disconnect()
 print(f'{client.user} left voice chat.')
 except Exception as e:

 print(f"Error during joining channel : {e}")
 else:
 print(f"Did not find a channel of ID {CHANNEL_ID}.")

client.run(TOKEN)


Any ideas ?


ffmpeg has been installed properly.


-
Why does File upload for moving image and Audio to tmp PHP folder work on Windows but only image upload portion works on Mac using MAMP ?
31 mai 2021, par YazdanSo according to my colleague who tested this on Windows says it works perfectly fine , but in my case when I use it on a Mac with MAMP for Moodle , the image files get uploaded to the correct destination folder without an issue whereas the audio files don't move from the tmp folder to the actual destination folder and to check if this was the case ... I just changed and gave a fixed path instead of
$fileTmpLocand the file made it to the correct destination. Sorry I know the first half of the code isn't the main issue but I still wanted to post the whole code so one could understand it easily, moreover I am just beginning to code so please "have a bit of patience with me" . Thanks in advance


// this file contains upload function 
// checks if the file exists in server
include("../db/database.php");
require_once(dirname(__FILE__) . '/../../../config.php');
global $IP;

$ajaxdata = $_POST['mediaUpload'];

$FILENAME = $ajaxdata[1];
$IMAGE=$ajaxdata[0];
// an array to check which category the media belongs too
$animal= array("bird","cat","dog","horse","sheep","cow","elephant","bear","giraffe","zebra");
$allowedExts = array("mp3","wav");
$temp = explode(".", $_FILES["audio"]["name"]);
$extension = end($temp);



$test = $_FILES["audio"]["type"]; 


if (
 $_FILES["audio"]["type"] == "audio/wav"||
 $_FILES["audio"]["type"] == "audio/mp3"||
 $_FILES["audio"]["type"] == "audio/mpeg"
 &&
 in_array($extension, $allowedExts)
 )
 {

 // if the name detected by object detection is present in the animal array
 // then initialize target path to animal database or to others
 if (in_array($FILENAME, $animal)) 
 { 
 $image_target_dir = "image_dir/";
 $audio_target_dir = "audio_dir/";
 } 
 else
 { 
 $image_target_dir = "other_image_dir/";
 $audio_target_dir = "other_audio_dir/";
 } 
 // Get file path
 
 $img = $IMAGE;
 // decode base64 image
 $img = str_replace('data:image/png;base64,', '', $img);
 $img = str_replace(' ', '+', $img);
 $image_data = base64_decode($img);

 //$extension = pathinfo( $_FILES["fileUpload"]["name"], PATHINFO_EXTENSION ); // jpg
 $image_extension = "png";
 $image_target_file =$image_target_dir . basename($FILENAME . "." . $image_extension);
 $image_file_upload = "http://localhost:8888/moodle310/blocks/testblock/classes/".$image_target_file;
 
 
 $audio_extension ="mp3";
 $audio_target_file= $audio_target_dir . basename($FILENAME. "." . $audio_extension) ;
 $audio_file_upload = "http://localhost:8888/moodle310/blocks/testblock/classes/".$audio_target_file;

 // file size limit
 if(($_FILES["audio"]["size"])<=51242880)
 {

 $fileName = $_FILES["audio"]["name"]; // The file name
 $fileTmpLoc = $_FILES["audio"]["tmp_name"]; // File in the PHP tmp folder
 $fileType = $_FILES["audio"]["type"]; // The type of file it is
 $fileSize = $_FILES["audio"]["size"]; // File size in bytes
 $fileErrorMsg = $_FILES["audio"]["error"]; // 0 for false... and 1 for true
 
 if (in_array($FILENAME, $animal)) 
 { 
 $sql = "INSERT INTO mdl_media_animal (animal_image_path,animal_name,animal_audio_path) VALUES ('$image_file_upload','$FILENAME','$audio_file_upload')";
 } else {
 $sql = "INSERT INTO mdl_media_others (others_image_path,others_name,others_audio_path) VALUES ('$image_file_upload','$FILENAME','$audio_file_upload')";
 }

 // if file exists
 if (file_exists($audio_target_file) || file_exists($image_target_file)) {
 echo "alert";
 } else {
 // write image file
 if (file_put_contents($image_target_file, $image_data) ) {
 // ffmpeg to write audio file
 $output = shell_exec("ffmpeg -i $fileTmpLoc -ab 160k -ac 2 -ar 44100 -vn $audio_target_file");
 echo $output;
 
 // $stmt = $conn->prepare($sql);
 $db = mysqli_connect("localhost", "root", "root", "moodle310"); 
 // echo $sql;
 if (!$db) {
 echo "nodb";
 die("Connection failed: " . mysqli_connect_error());
 }
 // echo"sucess";
 if(mysqli_query($db, $sql)){
 // if($stmt->execute()){
 echo $fileTmpLoc;
 echo "sucess"; 
 echo $output;
 }
 else {
 // echo "Error: " . $sql . "<br />" . mysqli_error($conn);
 echo "failed";
 }

 }else {
 echo "failed";
 }

 
 
 
 }
 
 // $test = "ffmpeg -i $outputfile -ab 160k -ac 2 -ar 44100 -vn bub.wav";
 } else
 {
 echo "File size exceeds 5 MB! Please try again!";
 }
}
else
{
 echo "PHP! Not a video! ";//.$extension." ".$_FILES["uploadimage"]["type"];
 }

?>


I am a student learning frontend but a project of mine requires a fair bit of backend. So forgive me if my question sounds silly.


What I meant by manually overriding it was creating another folder and a index.php file with
echo "hello"; $output = shell_exec("ffmpeg -i Elephant.mp3 -ab 160k -ac 2 -ar 44100 -vn bub.mp3"); echo $output;so only yes in this caseElephant.mp3was changed as the initial tmp path so in this case as suggested by Mr.CBroe the permissons shouldn't be an issue.

Okay I checked my
Apache_error.logonly to find out ffmpeg is indeed the culprit ... I had installedffmpegglobally so I am not sure if it is an access problem but here is a snippet of the log

I checked my php logs and found out that
FFmpegis the culprit.
Attached is a short log file

[Mon May 31 18:11:33 2021] [notice] caught SIGTERM, shutting down
[Mon May 31 18:11:40 2021] [notice] Digest: generating secret for digest authentication ...
[Mon May 31 18:11:40 2021] [notice] Digest: done
[Mon May 31 18:11:40 2021] [notice] Apache/2.2.34 (Unix) mod_ssl/2.2.34 OpenSSL/1.0.2o PHP/7.2.10 configured -- resuming normal operations
sh: ffmpeg: command not found
sh: ffmpeg: command not found
sh: ffmpeg: command not found


-
How to simultaneously capture mic, stream it to RTSP server and play it on iPhone's speaker ?
24 août 2021, par Norbert TowiańskiI want to capture sound from mic, stream it to RTSP server and play it simultaneously on iPhone's speaker after getting samples from RTSP server. I mean such kind of loop. I use FFMPEGKit and I want to use MobileVLCKit, but unfortunately microphone is off when I start play stream.
I think I've done first step (capturing from microphone and send OutputStream to RTSP server) :


@IBAction func transmitBtnPressed(_ sender: Any) {
 ffmpeg_transmit()
}

@IBAction func recordBtnPressed(_ sender: Any) {
 switch recordingState {
 case .idle:
 recordingState = .start
 startRecording()
 recordBtn.setTitle("Started", for: .normal)
 let urlToFile = URL(fileURLWithPath: outPipePath!)
 outputStream = OutputStream(url: urlToFile, append: false)
 outputStream!.open()
 case .capturing:
 recordingState = .end
 stopRecording()
 recordBtn.setTitle("End", for: .normal)
 default:
 break
 }
}

override func viewDidLoad() {
 super.viewDidLoad()
 outPipePath = FFmpegKitConfig.registerNewFFmpegPipe()
 self.setup()
}

override func viewDidAppear(_ animated: Bool) {
 super.viewDidAppear(animated)
 setUpAuthStatus()
}

func setUpAuthStatus() {
 if AVCaptureDevice.authorizationStatus(for: AVMediaType.audio) != .authorized {
 AVCaptureDevice.requestAccess(for: AVMediaType.audio, completionHandler: { (authorized) in
 DispatchQueue.main.async {
 if authorized {
 self.setup()
 }
 }
 })
 }
}

func setup() {
 self.session.sessionPreset = AVCaptureSession.Preset.high
 
 self.recordingURL = URL(fileURLWithPath: "\(NSTemporaryDirectory() as String)/file.m4a")
 if self.fileManager.isDeletableFile(atPath: self.recordingURL!.path) {
 _ = try? self.fileManager.removeItem(atPath: self.recordingURL!.path)
 }
 
 self.assetWriter = try? AVAssetWriter(outputURL: self.recordingURL!,
 fileType: AVFileType.m4a)
 self.assetWriter!.movieFragmentInterval = CMTime.invalid
 self.assetWriter!.shouldOptimizeForNetworkUse = true
 
 let audioSettings = [
 AVFormatIDKey: kAudioFormatLinearPCM,
 AVSampleRateKey: 48000.0,
 AVNumberOfChannelsKey: 1,
 AVLinearPCMIsFloatKey: false,
 AVLinearPCMBitDepthKey: 16,
 AVLinearPCMIsBigEndianKey: false,
 AVLinearPCMIsNonInterleaved: false,
 
 ] as [String : Any]
 
 
 self.audioInput = AVAssetWriterInput(mediaType: AVMediaType.audio,
 outputSettings: audioSettings)
 
 self.audioInput?.expectsMediaDataInRealTime = true
 
 if self.assetWriter!.canAdd(self.audioInput!) {
 self.assetWriter?.add(self.audioInput!)
 }
 
 self.session.startRunning()
 
 DispatchQueue.main.async {
 self.session.beginConfiguration()
 
 self.session.commitConfiguration()
 
 let audioDevice = AVCaptureDevice.default(for: AVMediaType.audio)
 let audioIn = try? AVCaptureDeviceInput(device: audioDevice!)
 
 if self.session.canAddInput(audioIn!) {
 self.session.addInput(audioIn!)
 }
 
 if self.session.canAddOutput(self.audioOutput) {
 self.session.addOutput(self.audioOutput)
 }
 
 self.audioConnection = self.audioOutput.connection(with: AVMediaType.audio)
 }
}

func startRecording() {
 if self.assetWriter?.startWriting() != true {
 print("error: \(self.assetWriter?.error.debugDescription ?? "")")
 }
 
 self.audioOutput.setSampleBufferDelegate(self, queue: self.recordingQueue)
}

func stopRecording() {
 self.audioOutput.setSampleBufferDelegate(nil, queue: nil)
 
 self.assetWriter?.finishWriting {
 print("Saved in folder \(self.recordingURL!)")
 }
}
func captureOutput(_ captureOutput: AVCaptureOutput, didOutput
 sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) {
 
 if !self.isRecordingSessionStarted {
 let presentationTime = CMSampleBufferGetPresentationTimeStamp(sampleBuffer)
 self.assetWriter?.startSession(atSourceTime: presentationTime)
 self.isRecordingSessionStarted = true
 recordingState = .capturing
 }
 
 var blockBuffer: CMBlockBuffer?
 var audioBufferList: AudioBufferList = AudioBufferList.init()
 
 CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(sampleBuffer, bufferListSizeNeededOut: nil, bufferListOut: &audioBufferList, bufferListSize: MemoryLayout<audiobufferlist>.size, blockBufferAllocator: nil, blockBufferMemoryAllocator: nil, flags: kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, blockBufferOut: &blockBuffer)
 let buffers = UnsafeMutableAudioBufferListPointer(&audioBufferList)
 
 for buffer in buffers {
 let u8ptr = buffer.mData!.assumingMemoryBound(to: UInt8.self)
 let output = outputStream!.write(u8ptr, maxLength: Int(buffer.mDataByteSize))
 
 if (output == -1) {
 let error = outputStream?.streamError
 print("\(#file) > \(#function) > Error on outputStream: \(error!.localizedDescription)")
 }
 else {
 print("\(#file) > \(#function) > Data sent")
 }
 }
}

func ffmpeg_transmit() {
 
 let cmd1: String = "-f s16le -ar 48000 -ac 1 -i "
 let cmd2: String = " -probesize 32 -analyzeduration 0 -c:a libopus -application lowdelay -ac 1 -ar 48000 -f rtsp -rtsp_transport udp rtsp://localhost:18556/mystream"
 let cmd = cmd1 + outPipePath! + cmd2
 
 print(cmd)
 
 ffmpegSession = FFmpegKit.executeAsync(cmd, withExecuteCallback: { ffmpegSession in
 
 let state = ffmpegSession?.getState()
 let returnCode = ffmpegSession?.getReturnCode()
 if let returnCode = returnCode, let get = ffmpegSession?.getFailStackTrace() {
 print("FFmpeg process exited with state \(String(describing: FFmpegKitConfig.sessionState(toString: state!))) and rc \(returnCode).\(get)")
 }
 }, withLogCallback: { log in
 
 }, withStatisticsCallback: { statistics in
 
 })
}
</audiobufferlist>

I want to use MobileVLCKit in that way :


func startStream(){
 guard let url = URL(string: "rtsp://localhost:18556/mystream") else {return}
 audioPlayer!.media = VLCMedia(url: url)

 audioPlayer!.media.addOption( "-vv")
 audioPlayer!.media.addOption( "--network-caching=10000")

 audioPlayer!.delegate = self
 audioPlayer!.audio.volume = 100

 audioPlayer!.play()

}


Could you give me some hints how to implement that ?


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